Presently, public switched telephone network (“PSTN”) land line telephony systems provide service utilizing centralized switching infrastructures which have knowledge as to each call termination end point type for a call. Based on this knowledge, this PSTN centralized switching infrastructure makes adjustments for end to end analog loss thereby optimizing performance with respect to acoustical audio level, audio distortion and echo. Guidance as to the audio loss across various end point types can be found in following standards: ANSI T1.508, “Network Performance—Loss Plan for Evolving Digital Networks” (“T1.508”); ANSI T1.401, “Network to Customer Installation Interfaces” (“T1.401”); TIA/EIA/TSB122A, “Telecommunications IP Telephony Equipment Voice Gateway Loss Level Plan Guidelines” (“TIA-122A”); and TIA/EIA/TIA-912, “Telecommunications IP Telephony Equipment Voice Gateway Transmission Requirements” (“TIA-912”).
Current Voice over Internet Protocol (“VoIP”) systems have adopted a fixed loss compensation approach at the VoIP end point based on these standards, and the assumption that the terminals at the two end points connected in a call will be of the same type (VoIP). This approach fails to recognize the current marketplace where the VoIP elements must, in fact, make connection to gateways for connection to PSTN land line services or to far end VoIP terminals which may not be compliant with the previously mentioned standards. This variability of far end terminals is not supported in current VoIP standards (PacketCable, ETSI, IETF or ITU), and with the industry movement to distributed call processing (SIP), the knowledge that historically has been available and resides in the PSTN infrastructure as to the type of far end connections is currently not available to the end points. Consequently, no information as to analog loss can be obtained and compensated for in response to such call connection variations.
Therefore, It would be advantageous to provide a system and method for dynamic end-to-end loss compensation in a VoIP system, with an ability to accommodate the characteristics of various types of terminals at the call endpoints. Furthermore, implementation of such utilizing Session Description Protocol (“SDP”), a protocol common to numerous IP telephony signaling schemes (Media Gateway Control Protocol—MGCP; Network-based Call Signaling—NCS; ITU H.323, etc.), would be desirable.